Hi, I am a "new member" trying to get up to speed with asterisk. Got 2 questions: 1. I have a setup at my shop, voip via voicenet.ca; asterisk 1.4 & aastra 480i terminal. No analog lines avail. It seems fine, except, for with some calls, I cannot have a dtmf tone recognised by the called pbx. (like press 0 for operator, etc.) On other pbx's it works OK.
2. I set up a client with two TDM400 & 6 POTS lines (only)& AAstra 9131 terminals. One application is to bridge an inbound call to another line & dial back out to a remote office. HOWEVER, the resulting call is at a very low volume & client is irritated. Changing the line parameters with fxotune just makes a bunch of clipping & echo, and one cannot tell which lines are to be bridged as it depends on the incoming call. Does anyone have insight? Should I (eat the TDM400s and) install sipura hardware? thanks Mark Borg
